Introduction

Modern Linux systems are more than capable of supporting your (semi-)professional audio needs. Latencies of 5ms down to even as low as 1ms can be achieved with good hardware and proper configuration. You could even run an industrial laser alongside your DAW (Digital audio workstation).

We believe Arch Linux provides competent users the right "framework" to sculpt their very own dedicated machines for recording, mixing, mastering, sound-design, DSP programming, and what have you. More often than not, when a "linux audio user" thinks of "sculpting", he thinks of "Gentoo". It is unknown what has led to such fallacious thinking, but needless to say, "compiling", "learning" and "performance" have as much to do with each other as "differences in rendering quality between Cubase and Pro Tools".

Pro Audio on Arch Linux

Binary-based

Arch is primarily a (meta-)distribution comprising binary packages, with a system for easy source-based package management as well. No need to waste time/CPU cycles, you just work™. This leads to "Third-party Repositores" below.

KISS

The philosophy of Arch requires packages to remain as vanilla as possible. You get what upstream software developers want you to get, including no-nonsense and straightforward package names with all headers. No more headaches due to missing development files when building a new Linux audio application.

Rolling-release

Arch is fast-paced in releasing updates and deprecating the old, following upstream schedules. Linux Audio is, to a certain extent, an experimental paradigm. Arch allows you to be up-to-date in order to keep up with your favourite audio software.

Pragmatic

Arch is agnostic with regards to licenses of software - it includes packages in the official repositories which "taint" the freedom of a distribution, but otherwise are popular and/or useful. This is also possible because Arch is not a complete "distribution", i.e releases are in fact only snapshots of the system with base packages only. License information is clearly available with every package without the need for installation, so it is up to the user to decide what he or she installs.

ABS

Arch provides a simple BSD Ports-like source packaging system, allowing the user to easily compile software via a buildscript ("recipe") and pass the resulting binary to the package manager. This way, any and all kinds of software, regardless of where they are from, can be monitored by the system. The number of new Linux audio software is always growing, so this is definitely a plus if the only way to get them is to build them from source.

AUR

Arch has an "unsupported" repository (AUR) of buildscripts open to all registered users for contribution, with thousands of ready-to-compile packages. Made for and by Arch Linux users. If something was released yesterday that would significantly help a number of users like you, chances are that today it has already been uploaded to the AUR.

Third-party Repositories

Arch in no way interrupts users when it comes to unofficial repositories that offer ready-to-run binary packages. It is simply a matter of adding one to the current list, and this is just in one file. No confusing "Whory Warthong", "Denim Jacket", "sources.list", or "source.keys" to worry about when you come across a new audio repository. We can afford to skip PEBKAC safeguards, because we expect users to know what they are doing. This means deciding on whether using a particular binary repository is "safe". For third-party Arch Linux binaries of audio software, we have this.

Q: So should I use Arch Linux (as my DAW)? I have been using Distro X so far..

A: If such is your question, we are sorry to inform you that the answer is negative.

Getting Started

Some of the major pro audio applications are already available from the official and community Arch Linux repositories. For anything which is not, you can either add a binary repository (see further down below) or if you prefer to compile, search the AUR. Nothing stops you from building directly off of upstream releases, but then you might as well run LFS.

Realtime configuration has mostly been automated. There is no longer any need to edit files like /etc/security/limits.conf for realtime access. However, if you must change the settings, see /etc/security/limits.d/99-audio.conf and /lib/udev/rules.d/40-hpet-permissions.rules (these files are provided by jack or jack2). Additionaly, you may want to increase the highest requested RTC interrupt frequency (default is 64 Hz) by adding to /etc/rc.local something like this:

By default, swap frequency defined by "swappiness" is set to 60. By reducing this number to 10, the system will wait much longer before trying to write to disk.
Then, there's inotify which watches for changes to files and reports them to applications requesting this information. When working with lots of audio data, a lot of watches will need to be kept track of, so they will need to be increased.
These two settings can be adjusted in /etc/sysctl.conf (file owned by procps).

vm.swappiness = 10
fs.inotify.max_user_watches = 524288

You may also want to maximize the PCI latency timer of the PCI sound card and raise the latency timer of all other PCI peripherals (default is 64).

See: Realtime for Users (Pay special attention especially if you do not run KDM, GDM or Slim.)

Have I rebooted after having done all that?

JACK

The aim here is to find the best possible combination of buffer size and periods, given the hardware you have. Frames/Period = 256 is a sane starter. For onboard and USB devices, try Periods/Buffer = 3. Commonly used values are: 256/3, 256/2, 128/3.

Also, the sample rate must match the hardware sample rate. Most often, 48000Hz is the common default across many of today's devices. Others include 44100 and 96000.

Almost always, when recording or sequencing with external gear is concerned, realtime is a must. Also, you may like to set maximum priority (at least 10 lower than system limits defined in /etc/security/limits.d/99-audio.conf); the highest is for the device itself).

And to check what sample and bit rates your device supports (for what samplerate it is set to, you have to look up the device manual or the knobs/switches/buttons on it):

$ cat /proc/asound/card0/codec#0

Replace card0 and codec#0 depending on what you have. You will be looking for rates or VRA in Extended ID.

Note: Once you set up JACK, try different audio applications to test your configuration results. I spent days trying to troubleshoot JACK xrun issues with LMMS which in the end turned out to be the problem with the latter.

JACK2

For jack2 you do not need qjackctl at all. You can use a lightweight alternative with less features. Applications like ardour or patchage already take care of client connections and smoothly adjusting the buffer size.

FireWire

Note: Nothing much is needed to be done as most things have been automated, especially with the introduction of the new FireWire stack, deprecation of HAL and more focus on udev. You should not need to edit device permissions, but if you suspect that your device may not be working due to such issues, see /lib/udev/rules.d/60-ffado.rules and if needed, create and put your changes into /etc/udev/rules.d/60-ffado.rules. Most often than not, your device will work with the libffado-svnAUR development version of the driver.

JACK(2) is built against FFADO, you only need to install it with the libffado package.

To test whether you have any chances of getting FireWire devices to work:

We cannot say for sure, particularly for those based on Ricoh (cross-platform issue). Most of the time, your device will run fine, but on occasion you will be faced with funny quirks. For unlucky ones, you will be facing hell.

A General Example

Realtime Kernel

Since a while ago, the stock Linux kernel has proven to be adequate for realtime uses. The stock kernel (with CONFIG_PREEMPT=y, default in Arch) can operate with a worst case latency of upto 10ms (time between the moment an interrupt occurs in hardware, and the moment the corresponding interrupt-thread gets running), although some device drivers can introduce latency much worse than that. So depending on your hardware and driver (and requirement), you might want a kernel with hard realtime capabilities.

The RT_PREEPMT patch by Ingo Molnar and Thomas Gleixner is an interesting option for hard and firm realtime applications, reaching from professional audio to industrial control.
Most audio-specific distro Linux ships with this patch applied. A realtime-preemptible kernel will also make it possible to tweak priorities of IRQ handling threads and help ensure smooth audio almost regardless of the load.

If you are going to compile your own kernel, remember that removing modules/options does not equate to a "leaner and meaner" kernel. It is true that the size of the kernel image is reduced, but in today's systems it is not as much of an issue as it was back in 1995.

In any way, you should also ensure that:

Timer Frequency is set to 1000Hz (CONFIG_HZ_1000=y; if you do not do MIDI you can ignore this)

APM is DISABLED (CONFIG_APM=n; Troublesome with some hardware - default in x86_64)

If you truly want a slim system, we suggest you go your own way and deploy one with static /devs. You should, however, set your CPU architecture. Selecting "Core 2 Duo" for appropriate hardware will allow for a good deal of optimisation, but not so much as you go down the scale.

General issue(s) with (realtime) kernels:

Hyperthreading (if you suspect, disable in BIOS)

There are ready-to-run/compile patched kernels available in the ABS and AUR.

ABS

You can use ABS to recompile linux with the patch. However, this is not the most useful of methods since updates will overwrite your custom kernel (at least you should add IgnorePkg=linux to /etc/pacman.conf).

MIDI

To work with MIDI you can it is highly recommended that you install a2j, a bridge between alsa midi and jack midi. It allows you to connect applications that only communicate with alsa midi to applications that only use jack midi. Laditray can also start/stop a2j.

Environment Variables

If you install things to non-standard directories, it is often necessary to set environment path variables so that applications know where to look (for plug-ins and other libraries). This usually affects only VST since users might have a Wine or external Windows location.

We would usually not have Linux plug-ins (LADSPA, LV2, DSSI, LXVST) beyond standard paths, so it is not necessary to export them. But if you do, be sure to include those standard paths as well since Arch does not do anything for dssi or ladspa, and some applications like dssi-vst will not look anywhere else if it finds predefined paths.

Tips and Tricks

IRQ issues can occur and cause problems. An example is video hardware reserving the bus, causing needless interrupts in the system I/O path. See discussion at FFADO IRQ Priorities How-To. If you have a realtime or a recent kernel, you can use this helpful script rtirq to adjust priorities of IRQ handling threads. Also available as systemd-rtirqAUR.

If you are facing a lot of xruns especially with nvidia, disable your GPU throttling. This can be done via the card's control applet and for nvidia it is "prefer maximum performance" (thanks to a mail in LAU by Frank Kober).

Hardware

M-Audio Delta 1010

The M-Audio Delta series cards are based on the VIA Ice1712 audio chipset.
Cards using this chip require that you install the alsa-tools package, because
it contains the envy24control program. Envy24control is a hardware level
mixer/controller. You can use alsa-mixer but you will save yourself some
hassle not to try it. Note that this section has no information on MIDI setup or
usage.

Open the mixer application:

$ envy24control

This application can be more than a bit confusing; see envy24control for guidance
on its use. That said, here is a very simple working setup for multitracking with Ardour.

On the "Monitor Inputs" and "Monitor PCMs" tabs, set all monitor inputs and monitor PCM's to around 20.

On the "Patchbay / Router" tab, set all to PCM out.

On the "Hardware Settings" tab, verify that the Master Clock setting matches what is set in Qjackctl. If these do not match you will have xruns out of control!

M-Audio Fast Track Pro

The M-Audio Fast Track Pro is an USB 4x4 audio interface, working at 24bit/96kHz. Due to limitation of USB 1, this device requires additional setup to get access to all its features. Device works in one of two configuration:

where vid and pid are vendor and product id for M-Audio Fast Track Pro, index is desired device number and device_setup is desired device setup. Possible values for device_setup are:

device modes

device_setup value

bit depth

frequency

analog output

digital output

analog input

digital input

IO mode

0x0

16 bit

48kHz

+

+

+

+

4x4

0x9

24 bit

48kHz

+

+

+

-

2x4

0x13

24 bit

48kHz

+

+

-

+

2x4

0x5

24 bit

96kHz

*

*

*

*

2x0 or 0x2

The 24 bit/96kHz mode is special: it provides all input/output, but you can open only one of 4 interfaces at a time. If you for example open output interface and then try to open second output or input interface, you will see error in kernel log:

cannot submit datapipe for urb 0, error -28: not enough bandwidth

which is perfectly normal, because this is USB 1 device and cannot provide enough bandwidth to support more than single (2 channel) destination/source of that quality at a time.

Depending on the value of index it will setup two devices: hwYYY:0 and hwYYY:1, which will contain available inputs and outputs. First device is most likely to contain analog output and digital input, while second one will contain analog input and digital output. To find out which devices are linked where and if they are setup correctly, you can check /proc/asound/cardYYY/stream{0,1} . Below is list of important endpoints that will help in correctly identifying card connections (it easy to mistake analog and digital input or output connections before you get used to the device):

Volume levels are hardware and routing can be done through QjackCtl, even with more cards linked together, this is not a problem.
The ffadomixer does not work with this card yet, hopefully in the future we can control more aspects of the card through a software interface like that.

Restricted Software

Steinberg's SDKs

It is very clear - we can distribute neither the VST nor the ASIO headers in binary package form. However, whenever you are building a program which would host Windows .dll VST plug-ins, check for the following hints (that do not require use of any SDK):

dssi-vst

fst

vestige

With that said, if you are building a program which would host native .so VST plug-ins, then there is no escape. For such cases, Arch yet again allows us to maintain a uniform local software database. We can "install" the SDK system-wide - you simply have to download it yourself and place it in the packaging directory.

Note: Steinberg does not forbid redistribution of resulting products, nor dictate what license they can be under. There are many GPL-licensed VST plug-ins. As such, distributing binary packages of software built with these restricted headers is not a problem, because the headers are simply buildtime dependencies.

Arch Linux Pro Audio Project

Yes, we have one. Think of "Planet CCRMA" or "Pro Audio Overlay", less the academic connotations of the former: ArchAudio.

What this means is that the repositories are add-ons, i.e you need to have a running, sane Arch Linux installation.

It is a relatively new effort although the initiative has been around since
2006/2007.